view libao2/ao_alsa.c @ 34259:a9bfa5a9fcac

ao_alsa: drop check for sys/asoundlib.h and ALSA version The ALSA output driver uses some parts of the ALSA API that were introduced after the switch to alsa/asoundlib.h, so checking for the deprecated header location is pointless as it has no chance of working anyway. Also, now that both ALSA output and input work with the 0.9 and 1.0 series, there is no need to keep checking for the ALSA version.

/*
 * ALSA 0.9.x-1.x audio output driver
 *
 * Copyright (C) 2004 Alex Beregszaszi
 *
 * modified for real ALSA 0.9.0 support by Zsolt Barat <[e-mail]>
 * additional AC-3 passthrough support by Andy Lo A Foe <[e-mail]>
 * 08/22/2002 iec958-init rewritten and merged with common init, zsolt
 * 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
 * 04/25/2004 printfs converted to mp_msg, Zsolt.
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#include <errno.h>
#include <sys/time.h>
#include <stdlib.h>
#include <stdarg.h>
#include <ctype.h>
#include <math.h>
#include <string.h>
#include <alloca.h>
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>

#include "config.h"
#include "subopt-helper.h"
#include "mixer.h"
#include "mp_msg.h"
#include "help_mp.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"

static const ao_info_t info =
{
    "ALSA-0.9.x-1.x audio output",
    "alsa",
    "Alex Beregszaszi, Zsolt Barat <[e-mail]>",
    "under development"
};

LIBAO_EXTERN(alsa)

static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
static snd_pcm_hw_params_t *alsa_hwparams;
static snd_pcm_sw_params_t *alsa_swparams;

static size_t bytes_per_sample;

static int alsa_can_pause;

#define ALSA_DEVICE_SIZE 256

static void alsa_error_handler(const char *file, int line, const char *function,
			       int err, const char *format, ...)
{
  char tmp[0xc00];
  va_list va;

  va_start(va, format);
  vsnprintf(tmp, sizeof tmp, format, va);
  va_end(va);
  tmp[sizeof tmp - 1] = '\0';

  if (err)
    mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
	   file, line, function, tmp, snd_strerror(err));
  else
    mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
	   file, line, function, tmp);
}

/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
  switch(cmd) {
  case AOCONTROL_QUERY_FORMAT:
    return CONTROL_TRUE;
  case AOCONTROL_GET_VOLUME:
  case AOCONTROL_SET_VOLUME:
    {
      ao_control_vol_t *vol = (ao_control_vol_t *)arg;

      int err;
      snd_mixer_t *handle;
      snd_mixer_elem_t *elem;
      snd_mixer_selem_id_t *sid;

      char *mix_name = "PCM";
      char *card = "default";
      int mix_index = 0;

      long pmin, pmax;
      long get_vol, set_vol;
      float f_multi;

      if(AF_FORMAT_IS_AC3(ao_data.format) || AF_FORMAT_IS_IEC61937(ao_data.format))
	return CONTROL_TRUE;

      if(mixer_channel) {
	 char *test_mix_index;

	 mix_name = strdup(mixer_channel);
	 if ((test_mix_index = strchr(mix_name, ','))){
		*test_mix_index = 0;
		test_mix_index++;
		mix_index = strtol(test_mix_index, &test_mix_index, 0);

		if (*test_mix_index){
		  mp_msg(MSGT_AO,MSGL_ERR,
		    MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
		  mix_index = 0 ;
		}
	 }
      }
      if(mixer_device) card = mixer_device;

      //allocate simple id
      snd_mixer_selem_id_alloca(&sid);

      //sets simple-mixer index and name
      snd_mixer_selem_id_set_index(sid, mix_index);
      snd_mixer_selem_id_set_name(sid, mix_name);

      if (mixer_channel) {
	free(mix_name);
	mix_name = NULL;
      }

      if ((err = snd_mixer_open(&handle, 0)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
	return CONTROL_ERROR;
      }

      if ((err = snd_mixer_attach(handle, card)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
	       card, snd_strerror(err));
	snd_mixer_close(handle);
	return CONTROL_ERROR;
      }

      if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err));
	snd_mixer_close(handle);
	return CONTROL_ERROR;
      }
      err = snd_mixer_load(handle);
      if (err < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, snd_strerror(err));
	snd_mixer_close(handle);
	return CONTROL_ERROR;
      }

      elem = snd_mixer_find_selem(handle, sid);
      if (!elem) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl,
	       snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
	snd_mixer_close(handle);
	return CONTROL_ERROR;
	}

      snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
      f_multi = (100 / (float)(pmax - pmin));

      if (cmd == AOCONTROL_SET_VOLUME) {

	set_vol = vol->left / f_multi + pmin + 0.5;

	//setting channels
	if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
		 snd_strerror(err));
	  snd_mixer_close(handle);
	  return CONTROL_ERROR;
	}
	mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);

	set_vol = vol->right / f_multi + pmin + 0.5;

	if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
		 snd_strerror(err));
	  snd_mixer_close(handle);
	  return CONTROL_ERROR;
	}
	mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
	       set_vol, pmin, pmax, f_multi);

	if (snd_mixer_selem_has_playback_switch(elem)) {
	  int lmute = (vol->left == 0.0);
	  int rmute = (vol->right == 0.0);
	  if (snd_mixer_selem_has_playback_switch_joined(elem)) {
	    lmute = rmute = lmute && rmute;
	  } else {
	    snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
	  }
	  snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
	}
      }
      else {
	snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
	vol->left = (get_vol - pmin) * f_multi;
	snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
	vol->right = (get_vol - pmin) * f_multi;

	mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
      }
      snd_mixer_close(handle);
      return CONTROL_OK;
    }

  } //end switch
  return CONTROL_UNKNOWN;
}

static void parse_device (char *dest, const char *src, int len)
{
  char *tmp;
  memmove(dest, src, len);
  dest[len] = 0;
  while ((tmp = strrchr(dest, '.')))
    tmp[0] = ',';
  while ((tmp = strrchr(dest, '=')))
    tmp[0] = ':';
}

static void print_help (void)
{
  mp_msg (MSGT_AO, MSGL_FATAL,
           MSGTR_AO_ALSA_CommandlineHelp);
}

static int str_maxlen(void *strp) {
  strarg_t *str = strp;
  return str->len <= ALSA_DEVICE_SIZE;
}

static int try_open_device(const char *device, int open_mode, int try_ac3)
{
  int err, len;
  char *ac3_device, *args;

  if (try_ac3) {
    /* to set the non-audio bit, use AES0=6 */
    len = strlen(device);
    ac3_device = malloc(len + 7 + 1);
    if (!ac3_device)
      return -ENOMEM;
    strcpy(ac3_device, device);
    args = strchr(ac3_device, ':');
    if (!args) {
      /* no existing parameters: add it behind device name */
      strcat(ac3_device, ":AES0=6");
    } else {
      do
	++args;
      while (isspace(*args));
      if (*args == '\0') {
	/* ":" but no parameters */
	strcat(ac3_device, "AES0=6");
      } else if (*args != '{') {
	/* a simple list of parameters: add it at the end of the list */
	strcat(ac3_device, ",AES0=6");
      } else {
	/* parameters in config syntax: add it inside the { } block */
	do
	  --len;
	while (len > 0 && isspace(ac3_device[len]));
	if (ac3_device[len] == '}')
	  strcpy(ac3_device + len, " AES0=6}");
      }
    }
    err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
		       open_mode);
    free(ac3_device);
    if (err >= 0)
      return err;
  }
  return snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK, open_mode);
}

/*
    open & setup audio device
    return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
    unsigned int alsa_buffer_time = 500000; /* 0.5 s */
    unsigned int alsa_fragcount = 16;
    int err;
    int block;
    strarg_t device;
    snd_pcm_uframes_t chunk_size;
    snd_pcm_uframes_t bufsize;
    snd_pcm_uframes_t boundary;
    const opt_t subopts[] = {
      {"block", OPT_ARG_BOOL, &block, NULL},
      {"device", OPT_ARG_STR, &device, str_maxlen},
      {NULL}
    };

    char alsa_device[ALSA_DEVICE_SIZE + 1];
    // make sure alsa_device is null-terminated even when using strncpy etc.
    memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);

    mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
	channels, format);
    alsa_handler = NULL;
#if SND_LIB_VERSION >= 0x010005
    mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
#else
    mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
#endif

    snd_lib_error_set_handler(alsa_error_handler);

    ao_data.samplerate = rate_hz;
    ao_data.format = format;
    ao_data.channels = channels;

    switch (format)
      {
      case AF_FORMAT_S8:
	alsa_format = SND_PCM_FORMAT_S8;
	break;
      case AF_FORMAT_U8:
	alsa_format = SND_PCM_FORMAT_U8;
	break;
      case AF_FORMAT_U16_LE:
	alsa_format = SND_PCM_FORMAT_U16_LE;
	break;
      case AF_FORMAT_U16_BE:
	alsa_format = SND_PCM_FORMAT_U16_BE;
	break;
      case AF_FORMAT_AC3_LE:
      case AF_FORMAT_S16_LE:
      case AF_FORMAT_IEC61937_LE:
	alsa_format = SND_PCM_FORMAT_S16_LE;
	break;
      case AF_FORMAT_AC3_BE:
      case AF_FORMAT_S16_BE:
      case AF_FORMAT_IEC61937_BE:
	alsa_format = SND_PCM_FORMAT_S16_BE;
	break;
      case AF_FORMAT_U32_LE:
	alsa_format = SND_PCM_FORMAT_U32_LE;
	break;
      case AF_FORMAT_U32_BE:
	alsa_format = SND_PCM_FORMAT_U32_BE;
	break;
      case AF_FORMAT_S32_LE:
	alsa_format = SND_PCM_FORMAT_S32_LE;
	break;
      case AF_FORMAT_S32_BE:
	alsa_format = SND_PCM_FORMAT_S32_BE;
	break;
      case AF_FORMAT_U24_LE:
	alsa_format = SND_PCM_FORMAT_U24_3LE;
	break;
      case AF_FORMAT_U24_BE:
	alsa_format = SND_PCM_FORMAT_U24_3BE;
	break;
      case AF_FORMAT_S24_LE:
	alsa_format = SND_PCM_FORMAT_S24_3LE;
	break;
      case AF_FORMAT_S24_BE:
	alsa_format = SND_PCM_FORMAT_S24_3BE;
	break;
      case AF_FORMAT_FLOAT_LE:
	alsa_format = SND_PCM_FORMAT_FLOAT_LE;
	break;
      case AF_FORMAT_FLOAT_BE:
	alsa_format = SND_PCM_FORMAT_FLOAT_BE;
	break;
      case AF_FORMAT_MU_LAW:
	alsa_format = SND_PCM_FORMAT_MU_LAW;
	break;
      case AF_FORMAT_A_LAW:
	alsa_format = SND_PCM_FORMAT_A_LAW;
	break;

      default:
	alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
	break;
      }

    //subdevice parsing
    // set defaults
    block = 1;
    /* switch for spdif
     * sets opening sequence for SPDIF
     * sets also the playback and other switches 'on the fly'
     * while opening the abstract alias for the spdif subdevice
     * 'iec958'
     */
    if (AF_FORMAT_IS_AC3(format) || AF_FORMAT_IS_IEC61937(format)) {
	device.str = "iec958";
	mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", channels);
    }
  else
        /* in any case for multichannel playback we should select
         * appropriate device
         */
        switch (channels) {
	case 1:
	case 2:
	  device.str = "default";
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
	  break;
	case 4:
	  if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
	    // hack - use the converter plugin
	    device.str = "plug:surround40";
	  else
	    device.str = "surround40";
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
	  break;
	case 6:
	  if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
	    device.str = "plug:surround51";
	  else
	    device.str = "surround51";
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
	  break;
	case 8:
	  if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
	    device.str = "plug:surround71";
	  else
	    device.str = "surround71";
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
	  break;
	default:
	  device.str = "default";
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
        }
    device.len = strlen(device.str);
    if (subopt_parse(ao_subdevice, subopts) != 0) {
        print_help();
        return 0;
    }
    parse_device(alsa_device, device.str, device.len);

    mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);

    if (!alsa_handler) {
      int open_mode = block ? 0 : SND_PCM_NONBLOCK;
      int isac3 =  AF_FORMAT_IS_AC3(format) || AF_FORMAT_IS_IEC61937(format);
      //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
      if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
	{
	  if (err != -EBUSY && !block) {
	    mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
	    if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
	      mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
	      return 0;
	    }
	  } else {
	    mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
	    return 0;
	  }
	}

      if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
      } else {
	mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
      }

      snd_pcm_hw_params_alloca(&alsa_hwparams);
      snd_pcm_sw_params_alloca(&alsa_swparams);

      // setting hw-parameters
      if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
		 snd_strerror(err));
	  return 0;
	}

      err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
					 SND_PCM_ACCESS_RW_INTERLEAVED);
      if (err < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
	       snd_strerror(err));
	return 0;
      }

      /* workaround for nonsupported formats
	 sets default format to S16_LE if the given formats aren't supported */
      if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
                                             alsa_format)) < 0)
      {
         mp_msg(MSGT_AO,MSGL_INFO,
		MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
         alsa_format = SND_PCM_FORMAT_S16_LE;
         if (AF_FORMAT_IS_AC3(ao_data.format))
           ao_data.format = AF_FORMAT_AC3_LE;
         else if (AF_FORMAT_IS_IEC61937(ao_data.format))
           ao_data.format = AF_FORMAT_IEC61937_LE;
         else
         ao_data.format = AF_FORMAT_S16_LE;
      }

      if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
					      alsa_format)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
		 snd_strerror(err));
	  return 0;
	}

      if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
						     &ao_data.channels)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
		 snd_strerror(err));
	  return 0;
	}

      /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
         prefer our own resampler, since that allows users to choose the resampler,
         even per file if desired */
#if SND_LIB_VERSION >= 0x010009
      if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
						     0)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
		 snd_strerror(err));
	  return 0;
	}
#endif

      if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
						 &ao_data.samplerate, NULL)) < 0)
        {
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
		 snd_strerror(err));
	  return 0;
        }

      bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
      bytes_per_sample *= ao_data.channels;
      ao_data.bps = ao_data.samplerate * bytes_per_sample;

	if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
							  &alsa_buffer_time, NULL)) < 0)
	  {
	    mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
		   snd_strerror(err));
	    return 0;
	  }

	if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
						      &alsa_fragcount, NULL)) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
		 snd_strerror(err));
	  return 0;
	}

      /* finally install hardware parameters */
      if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
		 snd_strerror(err));
	  return 0;
	}
      // end setting hw-params


      // gets buffersize for control
      if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
	  return 0;
	}
      else {
	ao_data.buffersize = bufsize * bytes_per_sample;
	  mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
      }

      if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
	return 0;
      } else {
	mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
      }
      ao_data.outburst = chunk_size * bytes_per_sample;

      /* setting software parameters */
      if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
	       snd_strerror(err));
	return 0;
      }
#if SND_LIB_VERSION >= 0x000901
      if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
	       snd_strerror(err));
	return 0;
      }
#else
      boundary = 0x7fffffff;
#endif
      /* start playing when one period has been written */
      if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
	       snd_strerror(err));
	return 0;
      }
      /* disable underrun reporting */
      if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
	       snd_strerror(err));
	return 0;
      }
#if SND_LIB_VERSION >= 0x000901
      /* play silence when there is an underrun */
      if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
	       snd_strerror(err));
	return 0;
      }
#endif
      if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
	       snd_strerror(err));
	return 0;
      }
      /* end setting sw-params */

      mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
	     ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
	     snd_pcm_format_description(alsa_format));

    } // end switch alsa_handler (spdif)
    alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
    return 1;
} // end init


/* close audio device */
static void uninit(int immed)
{

  if (alsa_handler) {
    int err;

    if (!immed)
      snd_pcm_drain(alsa_handler);

    if ((err = snd_pcm_close(alsa_handler)) < 0)
      {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err));
	return;
      }
    else {
      alsa_handler = NULL;
      mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
    }
  }
  else {
    mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined);
  }
}

static void audio_pause(void)
{
    int err;

    if (alsa_can_pause) {
        if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
        {
            mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, snd_strerror(err));
            return;
        }
          mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
    } else {
        if ((err = snd_pcm_drop(alsa_handler)) < 0)
        {
            mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err));
            return;
        }
    }
}

static void audio_resume(void)
{
    int err;

    if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
        mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
        while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
    }
    if (alsa_can_pause) {
        if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
        {
            mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, snd_strerror(err));
            return;
        }
          mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
    } else {
        if ((err = snd_pcm_prepare(alsa_handler)) < 0)
        {
           mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
            return;
        }
    }
}

/* stop playing and empty buffers (for seeking/pause) */
static void reset(void)
{
    int err;

    if ((err = snd_pcm_drop(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
	return;
    }
    if ((err = snd_pcm_prepare(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
	return;
    }
    return;
}

/*
    plays 'len' bytes of 'data'
    returns: number of bytes played
    modified last at 29.06.02 by jp
    thanxs for marius <[e-mail]> for giving us the light ;)
*/

static int play(void* data, int len, int flags)
{
  int num_frames;
  snd_pcm_sframes_t res = 0;
  if (!(flags & AOPLAY_FINAL_CHUNK))
      len = len / ao_data.outburst * ao_data.outburst;
  num_frames = len / bytes_per_sample;

  //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);

  if (!alsa_handler) {
    mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError);
    return 0;
  }

  if (num_frames == 0)
    return 0;

  do {
    res = snd_pcm_writei(alsa_handler, data, num_frames);

      if (res == -EINTR) {
	/* nothing to do */
	res = 0;
      }
      else if (res == -ESTRPIPE) {	/* suspend */
	mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
	while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
	  sleep(1);
      }
      if (res < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res));
	mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard);
	if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res));
	  return 0;
	  break;
	}
      }
  } while (res == 0);

  return res < 0 ? res : res * bytes_per_sample;
}

/* how many byes are free in the buffer */
static int get_space(void)
{
    snd_pcm_status_t *status;
    int ret;

    snd_pcm_status_alloca(&status);

    if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
    {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
	return 0;
    }

    ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
    if (ret > ao_data.buffersize)  // Buffer underrun?
	ret = ao_data.buffersize;
    return ret;
}

/* delay in seconds between first and last sample in buffer */
static float get_delay(void)
{
  if (alsa_handler) {
    snd_pcm_sframes_t delay;

    if (snd_pcm_delay(alsa_handler, &delay) < 0)
      return 0;

    if (delay < 0) {
      /* underrun - move the application pointer forward to catch up */
#if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
      snd_pcm_forward(alsa_handler, -delay);
#endif
      delay = 0;
    }
    return (float)delay / (float)ao_data.samplerate;
  } else {
    return 0;
  }
}